Communication networks, such as wide area networks (WAN), are commonly known, and perhaps the fastest growing of these is the Internet. One Internet application, known as multimedia transceiver, enables users to transmit and receive audio, video and text over the Internet. An example of this application, known as Internet telephony client, allows for telephone calls over the Internet.
In accordance with this Internet telephony client, a user dials the telephone number of a recipient user via a computer keyboard or the like. When the call is established between the users, the Internet telephony client digitally samples the voice of one user, temporary stores the samples in a buffer, and packages the samples into a data packet or packets. The data packet or packets is/are transmitted to a recipient user using an IP protocol. The recipient user receives the data packet or packets, strips them of the protocol headers and converts the samples into voice. This method is also performed at the caller end of the internet connection pathway.
This Internet telephony client exhibits drawbacks in that the voice quality on both ends of the communication pathway is poor. Several methods have been attempted to improve this voice quality.
At the sender end, these methods typically involve always transmitting packets that are built based on the parameters that are necessary for insuring the audio quality at the receiving side. These parameters can be redundancy, packaging schemes and/or patterns and compression type and/or rate. These methods exhibited drawbacks in that they did not adjust for changing network conditions, whereby increases in the network load continued to result in poor audio quality.
Moreover, when video communications were added to these already poor quality audio communications, the network load increased. This increase resulted in further delay noise and disturbance in audio, lowering its quality, and freezing of the video image.